Audio systems capable of reproducing digital format signals allow high fidelity sound and theater like effects compared to audio systems that can reproduce analog format signals. New digital audio disk formats such as super audio compact disk (SACD) allow reproduction of an extended range of frequencies compared to that of the more conventional digital compact disk (CD). The SACD contains a DSD signal while CDs contain PCM signals. Each sample of an analog audio signal is translated into either of the two binary digits 0 or 1 in a DSD signal encoder. The number of 0 and 1 binary digits over a given period of time determines the value of the analog audio signal.
Each PCM audio signal sample on a CD that represents a value of the analog audio signal is 16 bits in length. PCM signals have a uniform time period between samples, with the rate of sampling varying from 4 KiloHertz (KHz) to 192 KHz. DSD signals are sampled at much higher rates such as 2.8224 MegaHertz (MHz). Thus, an audio signal sampled using a PCM system at 44.1 KHz sampling rate and a DSD system at 2.8224 MHz sampling rate would result in 64 DSD samples occurring between each PCM sample.
Sampling an analog audio signal into a DSD signal may be performed by a sigma-delta modulator. A sigma-delta modulator includes analog circuitry that captures the analog audio signal and converts it into a single bit stream. Because of its single bit format, DSD signals can be converted back into the analog audio signal using minimal hardware. However, manipulation of the single bit DSD stream representing the analog audio signal can be very difficult. For example, tasks such as increasing the volume, adjusting treble or bass, etc., is very difficult because the DSD signal cannot be easily processed using existing digital filters and digital signal processing techniques. One solution is to convert the DSD signal into a PCM signal using a Finite Impulse Response (FIR) digital filter. PCM signals can be processed using digital filters and DSP techniques to allow manipulation of the PCM signal to accomplish tasks such as increasing the volume or adjusting bass and for more complex tasks such as surround sound effects. After manipulation of the PCM signal, the signal may be converted back to a DSD signal and/or into analog audio signal format for output to speakers.
Conversion of the DSD signal to a PCM signal may require a high quality and expensive FIR digital filter containing large quantities of complex hardware. The DSD to PCM converter may have an odd sized binary multiplier for multiplying 1 bit by the number of bits needed to encode the PCM signal (i.e. 1 by 16 bit multiplier, 1 by 24 bit multiplier, 1 by 32 bit multiplier, etc.). The DSD to PCM converter may also include a sign controller and an N-coefficient buffer to implement the FIR filter.
Digital signal processors (DSPs) that do not contain the dedicated hardware described above for DSD to PCM conversion are not capable of efficiently performing this conversion. Thus, there has been a longfelt need for an improved and low-cost method implemented in software or firmware and apparatus for efficient conversion of DSD signals to PCM signals in a (DSP).